Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

api/datachannelinterface.h

@@ -0,0 +1,183 @@
+/*
+ *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains interfaces for DataChannels
+// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel
+
+#ifndef WEBRTC_API_DATACHANNELINTERFACE_H_
+#define WEBRTC_API_DATACHANNELINTERFACE_H_
+
+#include <string>
+
+#include "webrtc/rtc_base/basictypes.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/copyonwritebuffer.h"
+#include "webrtc/rtc_base/refcount.h"
+
+namespace webrtc {
+
+// C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelinit
+// TODO(deadbeef): Use rtc::Optional for the "-1 if unset" things.
+struct DataChannelInit {
+  // Deprecated. Reliability is assumed, and channel will be unreliable if
+  // maxRetransmitTime or MaxRetransmits is set.
+  bool reliable = false;
+
+  // True if ordered delivery is required.
+  bool ordered = true;
+
+  // The max period of time in milliseconds in which retransmissions will be
+  // sent. After this time, no more retransmissions will be sent. -1 if unset.
+  //
+  // Cannot be set along with |maxRetransmits|.
+  int maxRetransmitTime = -1;
+
+  // The max number of retransmissions. -1 if unset.
+  //
+  // Cannot be set along with |maxRetransmitTime|.
+  int maxRetransmits = -1;
+
+  // This is set by the application and opaque to the WebRTC implementation.
+  std::string protocol;
+
+  // True if the channel has been externally negotiated and we do not send an
+  // in-band signalling in the form of an "open" message. If this is true, |id|
+  // below must be set; otherwise it should be unset and will be negotiated
+  // in-band.
+  bool negotiated = false;
+
+  // The stream id, or SID, for SCTP data channels. -1 if unset (see above).
+  int id = -1;
+};
+
+// At the JavaScript level, data can be passed in as a string or a blob, so
+// this structure's |binary| flag tells whether the data should be interpreted
+// as binary or text.
+struct DataBuffer {
+  DataBuffer(const rtc::CopyOnWriteBuffer& data, bool binary)
+      : data(data),
+        binary(binary) {
+  }
+  // For convenience for unit tests.
+  explicit DataBuffer(const std::string& text)
+      : data(text.data(), text.length()),
+        binary(false) {
+  }
+  size_t size() const { return data.size(); }
+
+  rtc::CopyOnWriteBuffer data;
+  // Indicates if the received data contains UTF-8 or binary data.
+  // Note that the upper layers are left to verify the UTF-8 encoding.
+  // TODO(jiayl): prefer to use an enum instead of a bool.
+  bool binary;
+};
+
+// Used to implement RTCDataChannel events.
+//
+// The code responding to these callbacks should unwind the stack before
+// using any other webrtc APIs; re-entrancy is not supported.
+class DataChannelObserver {
+ public:
+  // The data channel state have changed.
+  virtual void OnStateChange() = 0;
+  //  A data buffer was successfully received.
+  virtual void OnMessage(const DataBuffer& buffer) = 0;
+  // The data channel's buffered_amount has changed.
+  virtual void OnBufferedAmountChange(uint64_t previous_amount) {}
+
+ protected:
+  virtual ~DataChannelObserver() {}
+};
+
+class DataChannelInterface : public rtc::RefCountInterface {
+ public:
+  // C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelstate
+  // Unlikely to change, but keep in sync with DataChannel.java:State and
+  // RTCDataChannel.h:RTCDataChannelState.
+  enum DataState {
+    kConnecting,
+    kOpen,  // The DataChannel is ready to send data.
+    kClosing,
+    kClosed
+  };
+
+  static const char* DataStateString(DataState state) {
+    switch (state) {
+      case kConnecting:
+        return "connecting";
+      case kOpen:
+        return "open";
+      case kClosing:
+        return "closing";
+      case kClosed:
+        return "closed";
+    }
+    RTC_CHECK(false) << "Unknown DataChannel state: " << state;
+    return "";
+  }
+
+  // Used to receive events from the data channel. Only one observer can be
+  // registered at a time. UnregisterObserver should be called before the
+  // observer object is destroyed.
+  virtual void RegisterObserver(DataChannelObserver* observer) = 0;
+  virtual void UnregisterObserver() = 0;
+
+  // The label attribute represents a label that can be used to distinguish this
+  // DataChannel object from other DataChannel objects.
+  virtual std::string label() const = 0;
+
+  // The accessors below simply return the properties from the DataChannelInit
+  // the data channel was constructed with.
+  virtual bool reliable() const = 0;
+  // TODO(deadbeef): Remove these dummy implementations when all classes have
+  // implemented these APIs. They should all just return the values the
+  // DataChannel was created with.
+  virtual bool ordered() const { return false; }
+  virtual uint16_t maxRetransmitTime() const { return 0; }
+  virtual uint16_t maxRetransmits() const { return 0; }
+  virtual std::string protocol() const { return std::string(); }
+  virtual bool negotiated() const { return false; }
+
+  // Returns the ID from the DataChannelInit, if it was negotiated out-of-band.
+  // If negotiated in-band, this ID will be populated once the DTLS role is
+  // determined, and until then this will return -1.
+  virtual int id() const = 0;
+  virtual DataState state() const = 0;
+  virtual uint32_t messages_sent() const = 0;
+  virtual uint64_t bytes_sent() const = 0;
+  virtual uint32_t messages_received() const = 0;
+  virtual uint64_t bytes_received() const = 0;
+
+  // Returns the number of bytes of application data (UTF-8 text and binary
+  // data) that have been queued using Send but have not yet been processed at
+  // the SCTP level. See comment above Send below.
+  virtual uint64_t buffered_amount() const = 0;
+
+  // Begins the graceful data channel closing procedure. See:
+  // https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7
+  virtual void Close() = 0;
+
+  // Sends |data| to the remote peer. If the data can't be sent at the SCTP
+  // level (due to congestion control), it's buffered at the data channel level,
+  // up to a maximum of 16MB. If Send is called while this buffer is full, the
+  // data channel will be closed abruptly.
+  //
+  // So, it's important to use buffered_amount() and OnBufferedAmountChange to
+  // ensure the data channel is used efficiently but without filling this
+  // buffer.
+  virtual bool Send(const DataBuffer& buffer) = 0;
+
+ protected:
+  virtual ~DataChannelInterface() {}
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_DATACHANNELINTERFACE_H_